About two years ago, Google brought a new communications initiative called WebRTC to the two best known Internet standards organizations. WebRTC is short for Web Real Time Communications and the intent is to enable complex real time communications of voice, video and data using web clients, web servers and related applications. Google has been advancing the work both through contributions to open source libraries and by contributions to standards organizations. As you may know, once work is accepted by standards organizations, lots of people can get involved, so this work is no longer strictly a Google initiative and has gained support and participation from many companies both large and small.
The breakdown of work between the standards organizations has played to the strengths of two of them. The Internet Engineering Task Force (IETF) is contributing Internet protocols to the work and the Worldwide Web Consortium (W3C) is preparing an application program interface (API) based on JavaScript.
By the second half of last year, the drumbeats promoting WebRTC sounded loudly and in recent weeks, there was an industry conference dedicated strictly to WebRTC, with more to come later this year. I spoke at the SIPNOC conference on a WebRTC panel a couple of months back and there was lots of interest from telecom industry participants who have been busy in recent years building out real time communications using the Session Initiation Protocol (SIP). Some articles have even touted WebRTC as the “savior” for the telecom industry, whereas other pundits have said that WebRTC is very high on the hype scale.
One of the goals of this blog will be to cut through marketing spin and look at what is really happening in the world of communications. In my view, WebRTC has no shortage of hype, but there is also real technical substance in the initiative and many companies are making serious investments in WebRTC, even though many of the technical elements are nascent and the standards are not yet baked. One key thing to keep in mind is that WebRTC is the latest attempt to bring real time multimedia communications into the web infrastructure and make it relatively easy for web developers to add real time communications to their applications, without having the master the intricacies of SIP. The telecom industry has made several attempts to integrate with web developers in the past five years, but the WebRTC initiative seems more promising, since it is centered on web protocols, not on telecom protocols, and much of the “plumbing” will be buried beneath the same kind of JavaScript APIs that web developers have been utilizing for many years.
If you want a deep dive into WebRTC on the technical side, I can recommend the book “WebRTC: “APIs and RTCWEB Protocols of the HTML5 Real-Time Web,” written by Alan Johnston and Dan Burnett. They have just released a second edition, which I have not read yet, but the first edition offered a good technical overview and a nice distillation of the many standards that are being extended or developed as part of the overall initiative. (Disclosure: I know Alan well from his work in the IETF and we are co-authors on a current Internet Draft.) Since this is open standards work, you can also dive even deeper and sign up for the various IETF and W3C standards lists if you want to fill up your mailbox with emails.
Circling back to the title of this post, will WebRTC truly be a new communications paradigm? In my view, it’s too early to tell, but stay tuned and hold on tight. This promises to be quite a ride.